What is OpenSIPS development?
OpenSIPS development involves configuring and extending the OpenSIPS SIP proxy/server for high-performance routing, load balancing, and SIP application logic. CelloIP builds OpenSIPS deployments handling 50,000+ concurrent registrations with failover, dispatcher, and presence modules for enterprises and carriers.
When should I use OpenSIPS vs Kamailio?
OpenSIPS has more expressive scripting and richer built-in modules for complex routing logic. Kamailio has a larger community and is widely used for carrier-grade SBC deployments. CelloIP works with both; OpenSIPS is preferred for MVNO and CPaaS routing, Kamailio for SBC roles.
What can OpenSIPS be used for?
OpenSIPS is used for SIP load balancing, registrar/proxy, SBC, presence server, NAT traversal, rate limiting, and SIP trunk management. It is ideal for MVNO, CPaaS, and carrier deployments that need high scalability.
Expert OpenSIPS Development & SBC Solutions
CelloIP engineers build carrier-grade OpenSIPS SIP proxies, Session Border Controllers, LCR routing engines, and fraud detection systems. From 1,000 to millions of concurrent SIP transactions — architected to scale.
route[MAIN] {
# NAT detection
if (nat_uac_test("19"))
fix_nated_contact();
# SIP scanner protection
if (!pike_check_req()) {
xlog("L_ALERT","Blocked $si
");
exit;
}
# Authenticate SIP requests
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
exit;
}
# Topology hiding (SBC)
if (!topology_hiding("UC")) {
sl_send_reply("500","Error");
exit;
}
# Load balance to media servers
if (!ds_select_dst("1","4")) {
send_reply("503","Unavailable");
exit;
}
t_relay();
}OpenSIPS Development Services
Every OpenSIPS use case — from enterprise SIP proxy to carrier-grade SBC handling millions of transactions.
SIP Proxy Development
High-performance SIP proxy on OpenSIPS 3.x handling millions of transactions per second. Stateless and stateful routing, registrar, location service, NAT traversal (nathelper + rtpengine), and presence server.
Session Border Controller (SBC)
Enterprise and carrier SBC on OpenSIPS with topology hiding, back-to-back user agent (B2BUA), codec filtering, IP whitelisting, SIP header manipulation, media anchoring via rtpengine, and DoS protection.
Load Balancer & Dispatcher
SIP load balancing with OpenSIPS dispatcher module. Round-robin, weighted, least-load, and call-hash algorithms. Health-check with OPTIONS ping, failover to secondary destinations, and real-time capacity tracking.
LCR & Routing Engine
Least Cost Routing engine using OpenSIPS lcr module. Multiple carriers, per-prefix routing tables, time-of-day routing, quality-based routing using real-time ASR/ACD metrics, and automatic carrier failover.
OpenSIPS Script Development
Custom OpenSIPS routing scripts using the native scripting language and MI (Management Interface). Complex conditional routing logic, custom dialplan transformations, header manipulation, and ACC logging.
Fraud Detection & Security
Real-time VoIP fraud prevention: call rate limiting (pike module), IP blacklisting, authentication enforcement, IRSF detection, concurrent call limits per account, and SIP scanner blocking.
Presence & IM Server
OpenSIPS-based SIP presence server with SUBSCRIBE/NOTIFY support, BLF (Busy Lamp Field), shared line appearance, and instant messaging via MESSAGE method — integrated with enterprise PBX systems.
OpenSIPS Provisioning & MI
Web-based OpenSIPS management via OpenSIPS-CP (Control Panel), custom MI interfaces, REST API overlays for dynamic routing table updates, and integration with billing and subscriber management systems.
OpenSIPS Module Expertise
OpenSIPS's power comes from its 150+ loadable modules. CelloIP engineers have deep experience with every module relevant to enterprise and carrier deployments — selecting the right combination is critical to performance and reliability.
nathelperNAT detection and Contact rewritingrtpengineMedia proxy and SRTP bridgingdispatcherLoad balancing with health-checklcrLeast Cost Routing tablespikeDoS/DDoS rate limitingpermissionsIP-based ACL enforcementb2b_entitiesB2BUA for SBC topology hidingregistrarSIP registration serviceusrlocUser location databaseauth_dbDatabase-backed digest authaccCall accounting and CDR generationdialogDialog tracking for billing/limitscachedb_redisRedis-backed caching for scaleproto_tlsTLS transport for SIP signallingScaling Architecture Tiers
OpenSIPS architectures are designed for your current load with headroom for 10x growth — no rearchitecting when you scale.
OpenSIPS Development FAQ
What is OpenSIPS and what is it used for?
OpenSIPS is an open-source SIP server used primarily as a SIP proxy, registrar, Session Border Controller (SBC), and routing engine. Unlike Asterisk or FreeSWITCH which handle media processing, OpenSIPS focuses on ultra-high-performance SIP signalling routing — capable of handling millions of SIP transactions per second on a single server. It is used by carriers, MVNOs, ITSPs, and large enterprises to route SIP traffic, protect SIP infrastructure, provide load balancing, and implement least-cost routing.
What is the difference between OpenSIPS and Kamailio?
OpenSIPS and Kamailio are both derived from the same SER (SIP Express Router) codebase and share similar architecture. The key differences are in module availability, scripting language nuances, and community/commercial support. OpenSIPS has OpenSIPS-CP (a full web management panel), better built-in support for B2BUA functionality, and the OpenSIPS MI interface. Kamailio has a larger module library and is often preferred for very high-performance bare-metal routing. CelloIP has deep expertise in both and selects based on your specific requirements.
Can OpenSIPS work with Asterisk or FreeSWITCH?
Yes. A very common architecture pairs OpenSIPS as the front-end SIP proxy/load balancer with Asterisk or FreeSWITCH as the back-end media application server. OpenSIPS handles inbound SIP routing, authentication, NAT traversal, and load distribution. Asterisk or FreeSWITCH handles media processing, IVR, conferencing, and application logic. This hybrid architecture combines the scalability of OpenSIPS with the feature richness of Asterisk/FreeSWITCH.
How does OpenSIPS handle NAT traversal?
OpenSIPS handles NAT traversal using the nathelper module (which detects NAT via Contact/Via header inspection and rewrites them with the public IP) combined with rtpengine (a high-performance RTP media proxy that relays media between NAT-ed endpoints). This combination allows SIP signalling and RTP media to traverse symmetric NATs reliably — essential for enterprise deployments where users are behind corporate firewalls.
Flexible Engagement
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Whether you need a simple SIP proxy or a multi-continent SBC cluster, CelloIP has the OpenSIPS expertise to architect and deliver it.